1. Field of the Invention
This invention generally relates to the field of electronic music and audio signal processing and, particularly, to a digital signal processor for providing timbral change in arbitrary audio signals as a function of the input amplitude of the signal being processed.
2. Description of the Prior Art
In the field of electronic music and audio recording it has long been an ambition to achieve two goals: Music that is synthesized or recorded with maximum realism and music that selectively includes special sounds and effects created by electronic and studio techniques. To achieve these goals, electronic musical instruments for imitating acoustic instruments (realism) and creating new sounds (effects) have proliferated. Signal processors have been developed to make these electronic instruments and recordings of any instruments sound more convincing and to extend the spectral vocabularies of these instruments and recordings.
While considerable headway has been made in various synthesis techniques, including analog synthesis using oscillators, filters, etc., and frequency modulation synthesis, the greatest realism has been attained by the technique of digitally recording small segments of sound for playback by a keyboard or other controller. This technique is called sampling and yields some very realistic sounds. However, this sampling technique has one very significant drawback: Unlike acoustic phenomena, the timbre of the sound is the same at all playback amplitudes. This results in uninteresting sounds that are less complex, controllable and expressive than the acoustic instruments they imitate. Similar problems occur to different degrees with other synthesis techniques.
To increase the realism of synthesized music, a number of signal processing techniques have been employed. Most of these processes, such as reverberation, were originally developed for the alteration of acoustic sounds during the recording process. When applied to synthesized waveforms, they helped increase the sonic complexity and made them more natural sounding. However, none of the existing devices are able to relate timbral variation to changes in loudness with any flexibility. This relationship is well understood to be critical to the accurate emulation of acoustic phenomena. This invention provides a means of relating these two parameters, the processed result being more realistic and interesting than the input.
A number of signal processing techniques have been developed for achieving greater variety, control and special effects in the sound generating and recording process. In addition to the realism mentioned above, these signal processors have sought to extend the spectrum of available sounds in interesting ways. Also, to a large extent many of the dynamic techniques of signal processing have been well investigated for special effects, including time/amplitude, time/frequency, and input/output amplitude. These processes include, reverberators, filters, compressors and so on. None of these devices have the property of relating the amplitude of the input to the timbre of the output in such a way as to add musically useful and controllable harmonics to the signal being processed.
There are two areas of prior art that have direct bearing upon the invention: the use of non-linear transformation in non- real-time mainframe computer synthesis and in real-time sine-wave based hardware synthesis. Non-linear transformation of audio for music synthesis via the use of look-up tables has been in common use in universities worldwide since the mid-1970's. The seminal work in this field was done by Marc LeBrun and Daniel Arfib and published in the Journal of the Audio Engineering Society, V.27, #4 & V.27 #10. The work described in these writings gives an overview of waveshaping and makes extensive use of Chebyshev polynomials. The work done in this area consists primarily of the distortion of sine waves in order to achieve new timbres in music synthesis. There was a particular focus on brass instrumental sounds, as evidenced by the work of James Beauchamp, (Computer Music Journal V.3,#3 Sept, 1979) and others.
Hardware synthesis exploiting the non-linearity of analog components has been employed in music to distort waveforms for many years. Research in this area was done by Richard Schaefer in 1970 and 1971 and published in the Journal of the Audio Engineering Society, V.18,#4 and V.19,#7. In this literature he discusses the equations employed to achieve predictable harmonic results when synthesizing sound. With a sine wave input and using Chebyshev polynomials to determine the non-linear components used on the output circuitry, different waveforms were synthesized for electronic organs. More recently, Ralph Deutsch has employed hardware lookup tables as a real-time variation of the earlier mainframe synthesis techniques (U.S. Pat. #4,300,432). The Deutsch patents differ from the work by LeBrun, Arfib et al only inasmuch as multiple sine waves rather than single sine waves are input into the look-up table to achieve the synthesis of the desired output.
The primary limitation of the above mentioned uses of non- linear transformation are their employment in synthesis environments that did not allow real-time arbitrary audio input. By embedding the look-up tables or non-linear analog components in the synthesis circuitry or software, distortion of audio signals from outside the synthesis system was rendered impossible.
The advantage of this invention lies in its capacity to accept and transform arbitrary audio input. This opens up the possibility of performing non-linear transformation upon acoustic signals. Also, original or modified audio signals produced by any synthesis technique can be processed by the waveshaper. It also enables the insertion of the waveshaping circuitry into various signal processor configurations. Thus, it can be included as part of the recording/mixdown process before or after other signal processors, such as compressors, reverberators and filters.